Philipp Hancke ce88f7fa87 add DTLSSrtpTransport/SrtpTransport integration test
which shows that a DtlsSrtpTransport can send and receive
from the SrtpTransport which extracts the key from its DTLS transport.

The SrtpTransport takes its keys from the DtlsSrtpTransport which
(by the way of encryption and decryption) ensures both sides agree
on the keys to use

BUG=webrtc:357776213

Change-Id: I605c6ae660eab5a53bef69bcf84d7e70a34d7be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365274
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43231}
2024-10-14 09:47:45 +00:00
2024-10-09 12:57:21 +00:00
2024-10-10 11:21:44 +00:00
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2024-07-11 20:26:16 +00:00
2024-10-11 19:39:28 +00:00
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2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2024-05-27 12:46:57 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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